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/*
* Copyright (c) 2018 Paul B Mahol
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
#include "filters.h"
#include "window_func.h"
typedef struct HilbertContext {
const AVClass *class;
int sample_rate;
int nb_taps;
int nb_samples;
int win_func;
float *taps;
int64_t pts;
} HilbertContext;
#define OFFSET(x) offsetof(HilbertContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption hilbert_options[] = {
{ "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
{ "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
{ "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=22051}, 11, UINT16_MAX, FLAGS },
{ "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=22051}, 11, UINT16_MAX, FLAGS },
{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
WIN_FUNC_OPTION("w", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
{NULL}
};
AVFILTER_DEFINE_CLASS(hilbert);
static av_cold int init(AVFilterContext *ctx)
{
HilbertContext *s = ctx->priv;
if (!(s->nb_taps & 1)) {
av_log(s, AV_LOG_ERROR, "Number of taps %d must be odd length.\n", s->nb_taps);
return AVERROR(EINVAL);
}
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
HilbertContext *s = ctx->priv;
av_freep(&s->taps);
}
static av_cold int query_formats(AVFilterContext *ctx)
{
HilbertContext *s = ctx->priv;
static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
int sample_rates[] = { s->sample_rate, -1 };
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE
};
int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts);
if (ret < 0)
return ret;
return ff_set_common_samplerates_from_list(ctx, sample_rates);
}
static av_cold int config_props(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
HilbertContext *s = ctx->priv;
float overlap;
int i;
s->taps = av_malloc_array(s->nb_taps, sizeof(*s->taps));
if (!s->taps)
return AVERROR(ENOMEM);
generate_window_func(s->taps, s->nb_taps, s->win_func, &overlap);
for (i = 0; i < s->nb_taps; i++) {
int k = -(s->nb_taps / 2) + i;
if (k & 1) {
float pk = M_PI * k;
s->taps[i] *= (1.f - cosf(pk)) / pk;
} else {
s->taps[i] = 0.f;
}
}
s->pts = 0;
return 0;
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *outlink = ctx->outputs[0];
HilbertContext *s = ctx->priv;
AVFrame *frame;
int nb_samples;
if (!ff_outlink_frame_wanted(outlink))
return FFERROR_NOT_READY;
nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts);
if (nb_samples <= 0) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
return 0;
}
if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
return AVERROR(ENOMEM);
memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float));
frame->pts = s->pts;
s->pts += nb_samples;
return ff_filter_frame(outlink, frame);
}
static const AVFilterPad hilbert_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_props,
},
};
const AVFilter ff_asrc_hilbert = {
.name = "hilbert",
.description = NULL_IF_CONFIG_SMALL("Generate a Hilbert transform FIR coefficients."),
.init = init,
.uninit = uninit,
.activate = activate,
.priv_size = sizeof(HilbertContext),
.inputs = NULL,
FILTER_OUTPUTS(hilbert_outputs),
FILTER_QUERY_FUNC(query_formats),
.priv_class = &hilbert_class,
};
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